Wireless networks are increasingly being used for accessing the Internet. Wireless packet data protocols such as Universal Mobile Telecommunications System (UMTS), General Packet Radio Service (GPRS), EDGE, WCDMA, Fixed Wireless, and 4G technologies were developed to facilitate the transmission of data packets over the wireless network.
The Internet is a global network connecting computers from government agencies, educational institutions, the military, and businesses from around the world. Data is transmitted over the Internet using data packets. The data packets are sent from a sender to a recipient over any one of a number of network connections between the sender and recipient. Unlike a switched network, no dedicated connection between the sender and recipient is established. In contrast, the packets are sent from the sender with an address associated with the recipient, such as an Internet Protocol address (IP address) over any one of a number of available paths between the sender and recipient. This IP addressing scheme is also used within the Wireless Internet, along with other specific wireless protocols.
The Wireless Internet is intended to provide access to the Internet in general, but there are also application clusters and application infrastructure within the Wireless Carrier networks. Most 2.5G/3G/4G/Fixed Wireless wireless operators provide some data services from within their carrier network boundaries, either as a “walled garden” or a hybrid/quasi walled garden created through content and service provider partnerships. These walled gardens are separated from the Internet by firewalls and generally have had content or access methods modified to match wireless access device capabilities. There are a number of data service technologies that will reside on the Internet or within these walled gardens, supporting applications and services infrastructures such as:                Portal Services        Gaming Services        Streaming Media Services        WAP Services        Instant Messaging Services        Multimedia Messaging Services        Personal Network Storage Services        Location based Services        
Multimedia content delivery via streaming and downloading is one of the key services most 2.5G/3G carriers want to offer. This service will deliver various types of content including text, voice, music, and video clips. This content may be user to user based, such as transmitting a picture and text from a camera phone. Additionally, news, sports clips, and short animation (vector graphics)/macromedia clips may also be delivered as multi-media services.
Typical multimedia application and services that carriers desire to offer over wireless include the following:                (1) Streaming Media (Audio and Video)—On demand content:                    One to One with one wireline source and one wireless access receiver;                        (2) Live Webcasting (Audio and Video)—Live content:                    One to Many with one wireline source and many wireless access receivers;                        (3) Conferencing (Audio and Video)—Live content:                    Many to Many with many wireless sources and many wireless access receivers; and                        (4) Multimedia Downloads—On demand and Scheduled:                    One to One with one wireline source and one wireless access receiver.                        
Streaming media servers are part of the application cluster that works with other applications infrastructure to provide multimedia content for services. Multimedia delivery over wireless involves a delivery mechanism (streaming/downloading) that is adapted to wireless access characteristics, network resource awareness, session characteristics and Quality of Service (QoS) negotiated for each session. Current adaptation techniques involve end to end packet exchange between the delivery servers and end-user client applications on wireless terminals and other devices connected to wireless modems. Streaming servers use the Real Time Streaming Protocol (RTSP), the Real Time Transport Protocol (RTP), and the Real Time Control Protocol (RTCP) to deliver streaming multimedia. RTSP is used to setup and teardown connections besides performing other control features. RTSP also provides remote control functionality to play and pause streams to the client. The RTP protocol is used for media transport. The desired data is sent over RTP to and from a client 10 and a server 12 which runs on top of the User Datagram Protocol (UDP) in most implementations as is shown in FIG. 1. RTCP is used to exchange reports between parties in session. RTCP provides feedback on the quality of data distribution, can be used to send reports, and for synchronization of different media streams, such as lip syncing.
Both active senders and receivers send the RTCP reports. In some cases RTCP is exchanged between receivers in order to know if a problem is local or global. Streaming servers also use RTCP reports for control of adaptive streaming. From receiver reports, the server understands jitter, packet loss and round trip delays that are useful in adapting the streaming rate (variable bit rate encoding) to deliver the media content. RTCP message exchange is limited to a small and known fraction of the session bandwidth, with the interval between 2 RTCP packets recommended to be greater than 5 seconds. Most streaming media servers spend some initial time before the actual streaming of data packets to gather user perceived network characteristics. Servers use this to start streaming with a particular encoding rate. During the middle of streaming, if the server supports the adaptive encoding/streaming rate feature, it will periodically exchange RTCP packets to change streaming rate to suit user throughput.
FIGS. 2, 3, and 4 show the wireless adaptation protocol overhead involved for the delivery of 3 different multimedia applications. FIG. 2 is the Streaming media on demand where the client 20 (receiver) requests RR the stream from the server 22 (sender) or the server 22 requests SR the stream from the client 20. FIG. 3 represents a live Webcast, where request streams from a server. The participants in a Webcast can be senders 34, receivers 30, 31 and 32, or both. As the number of participants increase, the number of RTCP reports also increases. FIG. 4 depicts audio-video conferencing, in which participants are senders 46 and receivers 40, 42 and 44, and significant numbers of RTCP reports are required. This RTSP, RTP, RTCP packet exchange utilizes wireless access bandwidth and is subject to the lossy nature of the wireless air interface as well as the mobility of the wireless end-user. Such end-to-end packet exchange between delivery servers and wireless terminals for adaptation is cumbersome under substantial varying throughput conditions as well as zero throughput conditions. Furthermore, exchanging reports between the sender and receiver over wireless access is costly and information gathered through such reports is not always real-time. The responsiveness for content delivery adaptation is slow when reports on access and session characteristics are exchanged over air interfaces, which results in a degraded service delivery.
The goal of delivering multimedia services for wireless users, coupled with the challenges of current adaptation techniques, has created a need for clear heuristics and a statistical analysis of multimedia traffic to provide intelligent network resource aware media delivery for peak and off-peak times (busy hour vs. idle time). These improvements can then allow servers (senders/streaming media servers) to control the streaming flow rate or to stop or suspend media delivery during poor radio conditions and obtain higher precedence over other applications.